Re: [asterisk-users] t1 cards
On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? 300 metres, right? (not 300 miles?) Why stop at T1? Go for E1 :) with the right kit at each end you ought to be able to get 2Mb/sec or more. (distance depending) Personally, I'd go for a technology that gave me Ethernet at each end - then it makes it much easier to mix voice and data - But using something like a sync. modem and line driver then you need a media converter of some sorts at each end which might bump up the cost - at the savings of the E1 card in the PC though. Last time I had bare copper to play with (a BT EPS8 circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was running CHDLC over the link and acting as nothing more than a dumb media converter to give me Ethernet at each end. This was 6 years ago though. Ah, Looks like the technology has improved somewhat: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5261 From the UK site: Or even: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424mid=4946 (same thing from the UK site:) http://www.blackbox.co.uk/solutions/display.asp?cs=dvhid=1doc=lb300a-r2tx=LANsx=Network%20Appliances You need a pair, obviously... Hm. US site is $305, UK £253. Rip-off Britain again by the looks of it As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if using a LAN extender, then they're not neeed at all... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t1 cards
yes, more than 300 meters (longer than copper based ethernet allows). Yes to E1, as I understand it, it's just a config change on many cards anyway. I'm specificly looking at pci based t1/e1 cards because I'm finding single port cards on ebay going for 100-200 usd. in some cases I may want to drive a channel bank at the far end, thus t1/e1. anyone have experience on how far these pci based cards will drive when wired back to back? Eric On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? 300 metres, right? (not 300 miles?) Why stop at T1? Go for E1 :) with the right kit at each end you ought to be able to get 2Mb/sec or more. (distance depending) Personally, I'd go for a technology that gave me Ethernet at each end - then it makes it much easier to mix voice and data - But using something like a sync. modem and line driver then you need a media converter of some sorts at each end which might bump up the cost - at the savings of the E1 card in the PC though. Last time I had bare copper to play with (a BT EPS8 circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was running CHDLC over the link and acting as nothing more than a dumb media converter to give me Ethernet at each end. This was 6 years ago though. Ah, Looks like the technology has improved somewhat: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5261 From the UK site: Or even: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424mid=4946 (same thing from the UK site:) http://www.blackbox.co.uk/solutions/display.asp?cs=dvhid=1doc=lb300a-r2tx=LANsx=Network%20Appliances You need a pair, obviously... Hm. US site is $305, UK £253. Rip-off Britain again by the looks of it As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if using a LAN extender, then they're not neeed at all... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 stage dialing and 484 address incomplete
Hi, If my memory serves me right, there was thread (in dev mailing list ?) explaining how we could implement 2 stages dialing with SIP endpoints: user dials 1234 then asterisk replies 484 Address Incomplete, then user dials 5678 then asterisk begins to treat extension 12345678 as if it had been dialed as a whole. With compliant hardphones, you could get you phone to display a short text invite between the series of digits. This improves user experience, when consulting Voicemail, or asking features that need a parameter to be set. Is my memory correct ? Has anyone a clue ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Improving the voice Quality,
All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? Any thoughts? -Thank you, -Jai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip clients for smart phones?
hi; curious does anyone know of a smart phone client that could connect to asterisk? thanks Mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
What's a smart phone? Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? thanks Mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
windows smart phone v 6.0 example htc shadow is what i have. It has wifi abilitys. mike On Oct 3, 2008, at 12:11 AM, Alex Balashov wrote: What's a smart phone? Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? thanks Mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to add Callee's name into Dial command ?
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include this kind of
line :
To: sip [EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like this instead
?
To: John Doe sip [EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED];user=phone
Regards
___
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Re: [asterisk-users] How to add Callee's name into Dial command ?
2008/10/3 Olivier [EMAIL PROTECTED]
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include this kind of
line :
To: sip [EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like this
instead ?
To: John Doe sip [EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED];user=phone
I must add I also tried without success :
exten = _123X, 1, Set(SIALPEERNAME=Doe)
Regards
___
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
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Re: [asterisk-users] t1 cards
On Fri, 3 Oct 2008, Eric Fort wrote: yes, more than 300 meters (longer than copper based ethernet allows). Yes to E1, as I understand it, it's just a config change on many cards anyway. I'm specificly looking at pci based t1/e1 cards because I'm finding single port cards on ebay going for 100-200 usd. in some cases I may want to drive a channel bank at the far end, thus t1/e1. anyone have experience on how far these pci based cards will drive when wired back to back? Looks like this is the thing then: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5362 Just over $1000 a pair... couple that with an OpenVox PRI card at one end, channel bank at the other, and off you go... Gordon Eric On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? 300 metres, right? (not 300 miles?) Why stop at T1? Go for E1 :) with the right kit at each end you ought to be able to get 2Mb/sec or more. (distance depending) Personally, I'd go for a technology that gave me Ethernet at each end - then it makes it much easier to mix voice and data - But using something like a sync. modem and line driver then you need a media converter of some sorts at each end which might bump up the cost - at the savings of the E1 card in the PC though. Last time I had bare copper to play with (a BT EPS8 circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was running CHDLC over the link and acting as nothing more than a dumb media converter to give me Ethernet at each end. This was 6 years ago though. Ah, Looks like the technology has improved somewhat: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5261 From the UK site: Or even: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424mid=4946 (same thing from the UK site:) http://www.blackbox.co.uk/solutions/display.asp?cs=dvhid=1doc=lb300a-r2tx=LANsx=Network%20Appliances You need a pair, obviously... Hm. US site is $305, UK £253. Rip-off Britain again by the looks of it As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if using a LAN extender, then they're not neeed at all... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.22 and 1.6.0 Released
Asterisk Development Team schrieb: [Release info] Did anyone notice bug #0013531 (http://bugs.digium.com/view.php?id=13531)? It seems that the hold logic / MOH logic in chan_sip is somehow broken in 1.6.0... Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include
this kind of line :
To: sip [EMAIL PROTECTED] mailto:[EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like
this instead ?
To: John Doe sip [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead
of) callee's number which would offer a double check for caller
which might be confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED] mailto:[EMAIL PROTECTED];user=phone
I must add I also tried without success :
exten = _123X, 1, Set(SIALPEERNAME=Doe)
Have you tried Set(CALLERID(name)=Doe)? This the normal method for
setting caller ID names in Asterisk.
___
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Re: [asterisk-users] sip clients for smart phones?
On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
On Fri, Oct 3, 2008 at 10:08 AM, Babcock, Michael Alex [EMAIL PROTECTED] wrote: curious does anyone know of a smart phone client that could connect to I'm the Nokia N95 can talk to asterisk. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
Babcock, Michael Alex wrote: windows smart phone v 6.0 example htc shadow is what i have. It has wifi abilitys. Googling for windows mobile sip yeilds a multitude of results. I'm sure one of them will point you in the right direction. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
On asterisk CLI type rtp debug...YOu should see if DTMF tones are sent in the RTP flow and the protocol used Regards On Thu, Oct 2, 2008 at 11:42 PM, Barton Fisher [EMAIL PROTECTED] wrote: How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or 'rfc2833'? And more importantly if they could be sending both? If I specify 'inband' should they honor that? Thanks, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t1 cards
without any other hardware than 2 bare ass pci based t1/e1 cards wired back to back how far can one go between them? additional hardware defeats the purpose. Eric On Fri, Oct 3, 2008 at 3:01 AM, Gordon Henderson [EMAIL PROTECTED][EMAIL PROTECTED] wrote: On Fri, 3 Oct 2008, Eric Fort wrote: yes, more than 300 meters (longer than copper based ethernet allows). Yes to E1, as I understand it, it's just a config change on many cards anyway. I'm specificly looking at pci based t1/e1 cards because I'm finding single port cards on ebay going for 100-200 usd. in some cases I may want to drive a channel bank at the far end, thus t1/e1. anyone have experience on how far these pci based cards will drive when wired back to back? Looks like this is the thing then: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5362 Just over $1000 a pair... couple that with an OpenVox PRI card at one end, channel bank at the other, and off you go... Gordon Eric On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? 300 metres, right? (not 300 miles?) Why stop at T1? Go for E1 :) with the right kit at each end you ought to be able to get 2Mb/sec or more. (distance depending) Personally, I'd go for a technology that gave me Ethernet at each end - then it makes it much easier to mix voice and data - But using something like a sync. modem and line driver then you need a media converter of some sorts at each end which might bump up the cost - at the savings of the E1 card in the PC though. Last time I had bare copper to play with (a BT EPS8 circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was running CHDLC over the link and acting as nothing more than a dumb media converter to give me Ethernet at each end. This was 6 years ago though. Ah, Looks like the technology has improved somewhat: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5261 From the UK site: Or even: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424mid=4946 (same thing from the UK site:) http://www.blackbox.co.uk/solutions/display.asp?cs=dvhid=1doc=lb300a-r2tx=LANsx=Network%20Appliances You need a pair, obviously... Hm. US site is $305, UK £253. Rip-off Britain again by the looks of it As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if using a LAN extender, then they're not neeed at all... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
On Fri, Oct 03, 2008 at 01:09:41PM +0300, michel freiha wrote: On asterisk CLI type rtp debug...YOu should see if DTMF tones are sent in the RTP flow and the protocol used That is: you'll see there rfc2833 digits (sent as rtp signalling). This naturally won't show any DTMF digits sent in-band (as part of the audio stream) DTMF != digit -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t1 cards
I would say miles. DSL limits for equiv bandwidth is around 3 miles if I recall correctly. j On Fri, 3 Oct 2008, Eric Fort wrote: without any other hardware than 2 bare ass pci based t1/e1 cards wired back to back how far can one go between them? additional hardware defeats the purpose. Eric On Fri, Oct 3, 2008 at 3:01 AM, Gordon Henderson [EMAIL PROTECTED][EMAIL PROTECTED] wrote: On Fri, 3 Oct 2008, Eric Fort wrote: yes, more than 300 meters (longer than copper based ethernet allows). Yes to E1, as I understand it, it's just a config change on many cards anyway. I'm specificly looking at pci based t1/e1 cards because I'm finding single port cards on ebay going for 100-200 usd. in some cases I may want to drive a channel bank at the far end, thus t1/e1. anyone have experience on how far these pci based cards will drive when wired back to back? Looks like this is the thing then: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5362 Just over $1000 a pair... couple that with an OpenVox PRI card at one end, channel bank at the other, and off you go... Gordon Eric On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? 300 metres, right? (not 300 miles?) Why stop at T1? Go for E1 :) with the right kit at each end you ought to be able to get 2Mb/sec or more. (distance depending) Personally, I'd go for a technology that gave me Ethernet at each end - then it makes it much easier to mix voice and data - But using something like a sync. modem and line driver then you need a media converter of some sorts at each end which might bump up the cost - at the savings of the E1 card in the PC though. Last time I had bare copper to play with (a BT EPS8 circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was running CHDLC over the link and acting as nothing more than a dumb media converter to give me Ethernet at each end. This was 6 years ago though. Ah, Looks like the technology has improved somewhat: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5261 From the UK site: Or even: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424mid=4946 (same thing from the UK site:) http://www.blackbox.co.uk/solutions/display.asp?cs=dvhid=1doc=lb300a-r2tx=LANsx=Network%20Appliances You need a pair, obviously... Hm. US site is $305, UK ?253. Rip-off Britain again by the looks of it As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if using a LAN extender, then they're not neeed at all... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
Hi, [snip[ This way, I'm hoping to display callee's name beside (or instead of) callee's number which would offer a double check for caller which might be confusing extensions, for instance. you can set callerid per-peer in sip.conf like: callerid='Jhon Doe' 1234 this should work autmagically ;) cheers -- Daniele Santi .o. [EMAIL PROTECTED] ..o () ascii ribbon campaign Linux User #415108 ooo /\ www.asciiribbon.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
Unfortunaltely, both callerid='Jhon Doe' 1234 and Set(CALLERID(name)=Doe) don't work as I would like them to. For both, caller's phone screen still displays callee's number, not callee's name. When looking at SIP messages, you can can see that, unlike caller's name which is included in INVITE Header From: field, INVITE Header To: remains filled with callee's number, not its name. To make myself clear, here is what I'm trying to do : when Alice is calling Bob (Alice --- Asterisk---Bob), I would like Bob's phone to display Alice's name (no problem, for that) but I would also like Alice's phone screen to display Bob's name (instead of Bob's number) My SIP hardphone is capable of displaying P-Asserted-Identity in outbound calls (not just inbound) but I couldn't find any way to teach Asterisk to fill this P-Asserted-Identity header : Alice Asterisk Bob - INVITE bob --INVITE Bob - TRYING with Callername set TRYING with Callername set RINGING with Callername set RINGING with Callername set Regards 2008/10/3 Mr Shunz [EMAIL PROTECTED] Hi, [snip[ This way, I'm hoping to display callee's name beside (or instead of) callee's number which would offer a double check for caller which might be confusing extensions, for instance. you can set callerid per-peer in sip.conf like: callerid='Jhon Doe' 1234 this should work autmagically ;) cheers -- Daniele Santi .o. [EMAIL PROTECTED] ..o () ascii ribbon campaign Linux User #415108 ooo /\ www.asciiribbon.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
[snip]
To make myself clear, here is what I'm trying to do : when Alice is calling
Bob (Alice --- Asterisk---Bob), I would like Bob's phone to
display Alice's name (no problem, for that) but I would also like Alice's
phone screen to display Bob's name (instead of Bob's number)
mmm ... this wasn't clear on your OP ...
so you need to show the CALLED name on the CALLER phone ...
My SIP hardphone is capable of displaying P-Asserted-Identity in outbound
calls (not just inbound) but I
couldn't find any way to teach Asterisk to fill this P-Asserted-Identity
header :
you can try with: (*** untested ***)
exten = _123X, 1, SIPAddHeader(P-Asserted-Identity:
'${CALLERID(name)' ${CALLERID(num))
cheers
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Re: [asterisk-users] How to add Callee's name into Dial command ?
Mr Shunz wrote:
[snip]
To make myself clear, here is what I'm trying to do : when Alice is calling
Bob (Alice --- Asterisk---Bob), I would like Bob's phone to
display Alice's name (no problem, for that) but I would also like Alice's
phone screen to display Bob's name (instead of Bob's number)
mmm ... this wasn't clear on your OP ...
so you need to show the CALLED name on the CALLER phone ...
My SIP hardphone is capable of displaying P-Asserted-Identity in outbound
calls (not just inbound) but I
couldn't find any way to teach Asterisk to fill this P-Asserted-Identity
header :
you can try with: (*** untested ***)
exten = _123X, 1, SIPAddHeader(P-Asserted-Identity:
'${CALLERID(name)' ${CALLERID(num))
Interesting idea - I can see that being very useful. I know on my old
SPPA-841s they would do that - but it was based on looking up the dialed
number in the internal directory (which I programmed using a perl script
in the asterisk server.) So, when I dialed 213, the name appeared that I
dialed, confirming I had the right person.
Unfortunately, I never have been able to get my Polycom SoundPoint IP
500 phones to do that.
I just tested the SIPAddHeader command given above - doesn't work with
the Polycom Soundpoint IP 500 that I tested with. (Even with the missing
'}' at the end that I fixed - still doesn't work.)
Good idea thought - anybody have any magic that might make that work?
-josiah
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Re: [asterisk-users] How to add Callee's name into Dial command ?
(Even with the missing '}' at the end that I fixed oops, always forget about those :'( btw, in our installations we provide a centralized xml phonebook (and phones that support it, mostly grandstream and thomson), and they (the phones) autmatically set the called name on the display, both for internal and external numbers ;) cheers -- Daniele Santi .o. [EMAIL PROTECTED] ..o () ascii ribbon campaign Linux User #415108 ooo /\ www.asciiribbon.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
Hi all,
after a little bit of googleing, seems that correct syntax is:
exten = _123X, 1,
SIPAddHeader(P-Asserted-Identity:${CALLERID(name)}
sip:${CALLERID(num)})
(notice the sip:${CALLERID(num)})
but, IIUC, this sets the header for *outbound* call to 123X number, so
don't know if CALLER
can see the header ... and actually, ${CALLERID(num|name)} it's the
CALLER name, so maybe
should be ${EXTEN} (i.e. the CALLED number) like
exten = _123X, 1, SIPAddHeader(P-Asserted-Identity:doe sip:${EXTEN})
but sorry, can't test by now...
cheers
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[asterisk-users] MWI with Siemens Gigaset S450IP
Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but not when a message in present in voicemail. With SIP debug I can see 481 Call Leg/Transaction Does Not Exist replies to NOTIFY announcing new messages. With previous firmware, I had 415 Unsupported Media if my memory is correct. Has anyone been any further ? 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ok message
Dear All, I have a DTMF problem with VOxBone, the company that provide us the DID numbers...Sometimes they sent us DTMF packets and sometimes not... VoxBone said asterisk is not sending back OK message to their Gateway that's why they are not sending us the DTMF packets...How to force Asterisk server to reply back by sending OK message? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
I think this is what you want: http://bugs.digium.com/view.php?id=8824
On Fri, Oct 3, 2008 at 4:21 AM, Olivier [EMAIL PROTECTED] wrote:
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include this kind of
line :
To: sip [EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like this
instead ?
To: John Doe sip [EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED];user=phone
Regards
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Re: [asterisk-users] How to add Callee's name into Dial command ?
2008/10/3 Mr Shunz [EMAIL PROTECTED] (Even with the missing '}' at the end that I fixed oops, always forget about those :'( btw, in our installations we provide a centralized xml phonebook (and phones that support it, mostly grandstream and thomson), and they (the phones) autmatically set the called name on the display, both for internal and external numbers ;) Are you of that ? I'm not 100% certain, but I think Thomson phones wouldn't query centralized directory for outbound calls. I think centralized directory is only queried when using Directory key. cheers -- Daniele Santi .o. [EMAIL PROTECTED] ..o () ascii ribbon campaign Linux User #415108 ooo /\ www.asciiribbon.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t1 cards
T1 is NOT DSL. Most T1 links you purchase now are brought into your building with a type of DSL conversion to extend the distance between repeaters/amplifiers. T1 is purely a digital signal. DSL converts the ones and zeros to audio(multiple tones to provide multi channels of data). A simple analogy is comparing a T1 to DSL as a serial port to a modem. Back in the old days before fiber, copper T1's between CO's had their repeaters placed aproximately 1 mile apart. Best case going T1 port to T1 port, I would not expect this to work reliably at distances greater than one mile or 1.6 km but that does depend on the quality of the cable also. But in my mind, I would be seriously concerned about lightening protection. I have been around telco's and privately owned facilities for a long time and see lightening to be a very serious issue in this scenerio. I have seen short distance copper replaced by fiber because of issues over time with lightening damage despite having proper telco grade protection. Lyle Jeff LaCoursiere wrote: I would say miles. DSL limits for equiv bandwidth is around 3 miles if I recall correctly. j On Fri, 3 Oct 2008, Eric Fort wrote: without any other hardware than 2 bare ass pci based t1/e1 cards wired back to back how far can one go between them? additional hardware defeats the purpose. Eric On Fri, Oct 3, 2008 at 3:01 AM, Gordon Henderson [EMAIL PROTECTED][EMAIL PROTECTED] wrote: On Fri, 3 Oct 2008, Eric Fort wrote: yes, more than 300 meters (longer than copper based ethernet allows). Yes to E1, as I understand it, it's just a config change on many cards anyway. I'm specificly looking at pci based t1/e1 cards because I'm finding single port cards on ebay going for 100-200 usd. in some cases I may want to drive a channel bank at the far end, thus t1/e1. anyone have experience on how far these pci based cards will drive when wired back to back? Looks like this is the thing then: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5362 Just over $1000 a pair... couple that with an OpenVox PRI card at one end, channel bank at the other, and off you go... Gordon Eric On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 2 Oct 2008, Eric Fort wrote: I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? 300 metres, right? (not 300 miles?) Why stop at T1? Go for E1 :) with the right kit at each end you ought to be able to get 2Mb/sec or more. (distance depending) Personally, I'd go for a technology that gave me Ethernet at each end - then it makes it much easier to mix voice and data - But using something like a sync. modem and line driver then you need a media converter of some sorts at each end which might bump up the cost - at the savings of the E1 card in the PC though. Last time I had bare copper to play with (a BT EPS8 circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was running CHDLC over the link and acting as nothing more than a dumb media converter to give me Ethernet at each end. This was 6 years ago though. Ah, Looks like the technology has improved somewhat: http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468mid=5261 From the UK site: Or even: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424mid=4946 (same thing from the UK site:) http://www.blackbox.co.uk/solutions/display.asp?cs=dvhid=1doc=lb300a-r2tx=LANsx=Network%20Appliances You need a pair, obviously... Hm. US site is $305, UK ?253. Rip-off Britain again by the looks of it As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if using a LAN extender, then they're not neeed at all... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE
Re: [asterisk-users] How to add Callee's name into Dial command ?
2008/10/3 Mr Shunz [EMAIL PROTECTED]
Hi all,
after a little bit of googleing, seems that correct syntax is:
exten = _123X, 1,
SIPAddHeader(P-Asserted-Identity:${CALLERID(name)}
sip:${CALLERID(num)})
(notice the sip:${CALLERID(num)})
This could make P-Asserted-Identity field appear in INVITE header.
Thanks for that !!
Unfortunately, this doesn't help much to force Asterisk to query caller's
callerid field in sip.conf .
but, IIUC, this sets the header for *outbound* call to 123X number, so
don't know if CALLER
can see the header ... and actually, ${CALLERID(num|name)} it's the
CALLER name, so maybe
should be ${EXTEN} (i.e. the CALLED number) like
exten = _123X, 1, SIPAddHeader(P-Asserted-Identity:doe sip:${EXTEN})
but sorry, can't test by now...
cheers
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Re: [asterisk-users] How to add Callee's name into Dial command ?
2008/10/3 Joe Pukepail [EMAIL PROTECTED]
I think this is what you want: http://bugs.digium.com/view.php?id=8824
Thanks : this one very interesting.
Bottom line is it doesn't work at the moment right ?
http://bugs.digium.com/view.php?id=8824
On Fri, Oct 3, 2008 at 4:21 AM, Olivier [EMAIL PROTECTED] wrote:
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include this kind
of line :
To: sip [EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like this
instead ?
To: John Doe sip [EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED];user=phone
Regards
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Re: [asterisk-users] MWI with Siemens Gigaset S450IP
Olivier wrote: 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? I don't believe there is any support for hook-flash style transfers over SIP in Asterisk; that key should be programmed to use standard SIP transfer methods, not DTMF emulation methods. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ok message
michel freiha schrieb: I have a DTMF problem with VOxBone, the company that provide us the DID numbers...Sometimes they sent us DTMF packets and sometimes not... VoxBone said asterisk is not sending back OK message to their Gateway that's why they are not sending us the DTMF packets...How to force Asterisk server to reply back by sending OK message? Why should Asterisk send OK if they didn't send the DTMF packet in the first place? SIP debug trace? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ok message
200 OK is a SIP response indicating the successful establishment of an INVITE transaction. I can think of no reason why you would not be sending a 200 OK to your provider unless you are failing to Answer() the call in your dial plan and are instead sending them early media (183 Session in Progress). A packet capture would be most helpful. michel freiha wrote: Dear All, I have a DTMF problem with VOxBone, the company that provide us the DID numbers...Sometimes they sent us DTMF packets and sometimes not... VoxBone said asterisk is not sending back OK message to their Gateway that's why they are not sending us the DTMF packets...How to force Asterisk server to reply back by sending OK message? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
2008/10/3 Joe Pukepail [EMAIL PROTECTED]
I think this is what you want: http://bugs.digium.com/view.php?id=8824
Thanks : this one very interesting.
Bottom line is it doesn't work at the moment right ?
http://bugs.digium.com/view.php?id=8824
On Fri, Oct 3, 2008 at 4:21 AM, Olivier [EMAIL PROTECTED] wrote:
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include this kind of
line :
To: sip [EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like this instead
?
To: John Doe sip [EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip
[EMAIL PROTECTED];user=phone\)
but I still got :
To: sip [EMAIL PROTECTED];user=phone
Regards
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why you people need this thing in dial command which can possible with
sip.conf callerid options
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Re: [asterisk-users] sip clients for smart phones?
i should have been more clear, thanks a windows mobile smart phone sip client. sorry about that On Oct 3, 2008, at 1:54 AM, randulo wrote: On Fri, Oct 3, 2008 at 10:08 AM, Babcock, Michael Alex [EMAIL PROTECTED] wrote: curious does anyone know of a smart phone client that could connect to I'm the Nokia N95 can talk to asterisk. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
Why you didn't read the whole thread before saying that is beyond me. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of satish patel Sent: October 3, 2008 12:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to add Callee's name into Dial command ? [Mark Hamilton] snip why you people need this thing in dial command which can possible with sip.conf callerid options ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delaytos=0x08 high throughput tos=0x04 high reliabilitytos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
On Fri, 3 Oct 2008, Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. Not quite RTP traffic, but iperf might help - if you have access to both ends of the link. You can set a bandwidth and packet size which might help you. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
thanks i'll give this a try mike On Oct 3, 2008, at 10:56 AM, Mark Hamilton wrote: This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
try using Frig.. it's a great client with an SIP client.. i tried it on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi-Fi... and i DO use it with my two Asterisk Servers.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 3 Oct 2008 15:33:52 -0400 Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though.Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ See how Windows Mobile brings your life together—at home, work, or on the go. http://clk.atdmt.com/MRT/go/msnnkwxp1020093182mrt/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
is it frig or fring? On Oct 3, 2008, at 11:49 AM, Tariq .. wrote: try using Frig.. it's a great client with an SIP client.. i tried it on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi- Fi... and i DO use it with my two Asterisk Servers.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 3 Oct 2008 15:33:52 -0400 Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users See how Windows Mobile brings your life together—at home, work, or on the go. See Now___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
I don't even see it anywhere on TouchFlo3D. I don't see where to even use it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] network monitoring - triggering a phone call in asterisk
I'd like to integrate asterisk into the notification portion of my network monitoring where a call is generated and a message is played when a host or service failed to be reachable (something where nagios could trigger an event in asterisk would rock!) How could this best be accomplished? Anyone know an existing solution? Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
i'm checking out fring right now On Oct 3, 2008, at 12:02 PM, Mark Hamilton wrote: I don't even see it anywhere on TouchFlo3D. I don't see where to even use it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network monitoring - triggering a phone call in asterisk
Use callfiles, monitor Manager for the appropriate connect or failure events, output those to a state file or report them via a domain socket service that a Nagios plugin can connect to and inquire. Eric Fort wrote: I'd like to integrate asterisk into the notification portion of my network monitoring where a call is generated and a message is played when a host or service failed to be reachable (something where nagios could trigger an event in asterisk would rock!) How could this best be accomplished? Anyone know an existing solution? Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
it is FRING i'm sorry for the mistype... www.fring.com AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Fri, 3 Oct 2008 12:00:16 -0800Subject: Re: [asterisk-users] sip clients for smart phones?is it frig or fring? On Oct 3, 2008, at 11:49 AM, Tariq .. wrote: try using Frig.. it's a great client with an SIP client.. i tried it on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi-Fi... and i DO use it with my two Asterisk Servers.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 3 Oct 2008 15:33:52 -0400 Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though.Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users See how Windows Mobile brings your life together—at home, work, or on the go. See Now___-- Bandwidth and Colocation Provided by http://www.api-digital.com --AstriCon 2008 - September 22 - 25 Phoenix, ArizonaRegister Now: http://www.astricon.netasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy _ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie.
[asterisk-users] OT: Re: sip clients for smart phones?
Tariq, fix your email client (it eats line breaks in text/plain). And please leave out the parts which are not relevant (list policy). Thanks. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: sip clients for smart phones?
i'm using Hotmail webmail.. so what is wrong with it? AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Fri, 3 Oct 2008 22:17:02 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT: Re: sip clients for smart phones? Tariq, fix your email client (it eats line breaks in text/plain). And please leave out the parts which are not relevant (list policy). Thanks. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
that's ok i found it On Oct 3, 2008, at 12:15 PM, Tariq .. wrote: it is FRING i'm sorry for the mistype... www.fring.com AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 3 Oct 2008 12:00:16 -0800 Subject: Re: [asterisk-users] sip clients for smart phones? is it frig or fring? On Oct 3, 2008, at 11:49 AM, Tariq .. wrote: try using Frig.. it's a great client with an SIP client.. i tried it on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi- Fi... and i DO use it with my two Asterisk Servers.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 3 Oct 2008 15:33:52 -0400 Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users See how Windows Mobile brings your life together—at home, work, or on the go. See Now___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy Want to do more with
Re: [asterisk-users] sip clients for smart phones?
Ah, I don't use the touchflow crap :) On mine on the today screen (you'll have to go to settings, today, items) Set internet telephony to on and you should see it on the home screen. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 4:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? I don't even see it anywhere on TouchFlo3D. I don't see where to even use it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network monitoring - triggering a phone call in asterisk
This may be what you're looking for: http://www.linuxjournal.com/content/custom-checks-and-notifications-nagios Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, October 03, 2008 4:13 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] network monitoring - triggering a phone call in asterisk Use callfiles, monitor Manager for the appropriate connect or failure events, output those to a state file or report them via a domain socket service that a Nagios plugin can connect to and inquire. Eric Fort wrote: I'd like to integrate asterisk into the notification portion of my network monitoring where a call is generated and a message is played when a host or service failed to be reachable (something where nagios could trigger an event in asterisk would rock!) How could this best be accomplished? Anyone know an existing solution? Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1-beta1 Now Available
The Asterisk.org development team has released Asterisk 1.6.1-beta1. While Asterisk 1.6.0 was in the beta and release candidate stage, the development team was also working on merging new things for Asterisk 1.6.1. Now that 1.6.0 has been released, the testing cycle for Asterisk 1.6.1 has begun. To see the list of new features in Asterisk 1.6.1, see the CHANGES file: http://svn.digium.com/view/asterisk/branches/1.6.1/CHANGES?view=markup For a full list of changes that have gone into the development of Asterisk 1.6.1 so far, see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.6.1-beta1/ChangeLog?view=markup To download Asterisk 1.6.1-beta1, visit the Digium downloads site: http://downloads.digium.com/pub/telephony/asterisk/ Please report all issues to the issue tracker, http://bugs.digium.com/. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF issues...
I am having a big problem with DTMF. I have a customer using an Asterisk 1.4.20.1 system with ZTDUMMY as the timing source. The problem is that when they dial into a conference bridge or IVR where they have to enter a code they always get an error. Either some numbers are duplicated or missing. They use Teliax for calls to the USA and Protel in Mexico. Both carriers have the same problem so I think Asterisk could be at fault here. I have tried using dtmfmode as auto, inband, info and rfc2833 but get the same result. I have tried ulaw, alaw and g729 as the codec for the calls but have the same problem. Could this be a timing issue? Since there are no zap channels we use ZTDUMMY for timing. The server usually runs only about 15 simultaneous calls so the load is not heavy on the processor. What is the best method to debug DTMF issues? Do I have to sniff the SIP packets? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Talk Off
I am having some issues with dtmf beeping while in conversation. This is a simple home setup using asterisk 1.4.21 and a grandsteam 286 ATA with a cordless phone. I have played with the relaxdtmf setting in sip.conf. Setting it to yes made it worse which makes me think that the issue is coming from asterisk and not the ata. And all the dtmf modes are rfc. Any one have any tips to further trouble shoot this? Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF issues...
Hey Carlos, What is the best method to debug DTMF issues? Do I have to sniff the SIP packets? The best method to debug DTMF issues depend on how you receive those DTMF digits. Assuming you can use SIP INFO for the DTMF, that means the DTMF digits are not really DTMF :-), that is, is not audio, with SIP INFO the digits will be received by Asterisk as part of the SIP signaling protocol and therefore you can easily spot them using sip debug peer myxpeer. If I were you I'd try that and see what I am really receiving before drawing any other conclusion. Moy -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: sip clients for smart phones?
Tariq .. schrieb: i'm using Hotmail webmail.. so what is wrong with it? See for yourself: http://lists.digium.com/pipermail/asterisk-users/2008-October/219531.html http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html Ironically you would think that *my* email client is broken while actually yours messes up the text/plain part. And in addition to that - You don't skip the irrelevant parts (e.g. my signature or the list footer) - You top-post - You violate email netiquette by not using a proper signature separator - You send a footer telling me about Windows Live which is totally unrelated. Even for free email accounts that's not acceptable any longer since there are free accounts without advertising. I could live with 1 or maybe 2 of these issues but 5 is a bit much. You didn't even notice these problems, so, ok, sorry for being rude. But for people who are used to email in ages it feels like a punch in the face. It's a real culture clash. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
All, Just an update on this. This turned out to be a bug in Cisco firewall. We ended up in upgrading the Firmware on the firewall. One thing I want to add, this was first time we used the fail over unit during peak time. In the whole process (failover, upgrade and failover back to active unit) was completely seamless. Did not had any down time, there was just a pause for just 1 second in the audio. I was very impressed. -Jai On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi [EMAIL PROTECTED] wrote: Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
BT3 (BackTrack) LiveCD is one of the best things out there, even has sipp built right in, as well as other great apps, utilities, and security auditing. I suggest everyone have a copy in their arsenal, and it is free of course. Thanks, Steve Totaro On Fri, Oct 3, 2008 at 10:09 PM, Jai Rangi [EMAIL PROTECTED] wrote: All, Just an update on this. This turned out to be a bug in Cisco firewall. We ended up in upgrading the Firmware on the firewall. One thing I want to add, this was first time we used the fail over unit during peak time. In the whole process (failover, upgrade and failover back to active unit) was completely seamless. Did not had any down time, there was just a pause for just 1 second in the audio. I was very impressed. -Jai On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi [EMAIL PROTECTED] wrote: Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] is DNS SRV enough for failover?
This is so wrong it's not funny. The caching of DNS SRV records acts in its favor when it comes to failover - the UAs already have the information they need in their resolver cache to perform the failover without having to make another DNS query. The TTL you need to worry about is that of the SIP registration - UAs will typically renew their SIP registration at the half-life of the TTL. A short SIP registration TTL will permit devices which are not actively placing calls to failover more quickly than they otherwise would. Of course, a balance must be struck between short SIP registration TTLs, and the amount of SIP registration traffic this generates. YMMV. CP On Tue, 2008-09-30 at 22:07 -0400, Alex Balashov wrote: Nhadie wrote: hi, i'm using DNS SRV for failover, i tried to test shutting the server down, sip client should still register on the other server but it did not. i'm using x-lite which i don't know if it's doing a srv query. does this mean SRV is not enough for failover? if a client has dns caching would this cause a problem? SRV records are DNS. DNS is cached. Ergo, SRV records are cached. Ergo, if they are cached excessively - either because the TTL is long, or in defiance of the TTL - it can cause a problem. No, DNS is not a good way to do real-time failover for anything. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
